eac3to output too loud after -slowdown

[bugmen0t]

I have a lot of PAL MKV files that I want to slowdown to 23.976...fps. When I try to do that with eac3to, I get an Audio Clipping warning and the message that negative gain will be applied. This negative gain is always different, ranging from 1 to 3.5dB. But after encoding en remuxing in a new MKV container, the clipping still is present because of the louder audio (cranking the volume up to 100%). I pulled the audio from a lower quality source which originated from the same audio source, but not at PAL-speed. It was as loud as the PAL version, but without any clipping.

Is there a way to keep the same audio level before encoding? And if not, keep the clipping when using 100% volume from happening?

Here's an example of an encoding:

H:\PAL>eac3to testPAL.ac3 testNTSC.ac3 -384 -slowdown
AC3, 5.1 channels, 1:37:15, 384kbps, 48kHz
Decoding with libav/ffmpeg...
Changing FPS from 25.000 to 23.976...
Remapping channels...
Encoding AC3 <384kbps> with libAften...
Creating file "testNTSC.ac3"...
Clipping detected, a 2nd pass will be necessary.
Starting 2nd pass...
Decoding with libav/ffmpeg...
Changing FPS from 25.000 to 23.976...
Remapping channels...
Encoding AC3 <384kbps> with libAften...
Applying -2.35dB gain...
Creating file "testNTSC.ac3"...
eac3to processing took 58 seconds.
Done.

Bonus question: Can I get higher quality audio after encoding by using -448 with the -slowdown option?

[Richard1485]

448 ought to preserve more quality than 384. If you're remuxing to MKV, you could keep the default value of 640. And unless you have a specific need for AC-3, you could avoid a lossy recompression altogether and use FLAC (or PCM if you don't care about file-size).

[hello_hello]

Are you sure the clipping is clipping and not glitches in the encoding/re-sampling process? If that's likely to happen, I wouldn't know why, but if eac3to is running a second pass to prevent clipping, I'd imagine there shouldn't be any. If you want to try a different method to see if anything changes, MeGUI's audio encoder can do PAL speedup/slowdown. It re-encodes via Avisynth.

I use foobar2000 for audio re-encoding myself. There's a DSP which can change the pitch/speed when encoding (SoundTouch), but I'm not sure how practical it'd be trying to do a PAL/NTSC conversion with it. I've never tried it.
For "normal" conversions I just re-encode multi-channel audio (mainly DTS) without altering the volume and I haven't noticed any clipping. Thinking about it, I'm not sure why eac3to would detect clipping if there's no downmixing going on. Unless the re-sampling does change the volume a little in some way? If you run the same encode without the slowdown included, is there still clipping detected?

Unless you need AC3, I'd go with a different format myself. Mostly I don't re-encode AC3, but given you are, AAC should allow you to use a lower bitrate without sacrificing any extra quality. I tend to use NeroAAC and it's default VBR quality mode (q5.0). Foobar2000 just finished converting a 5.1ch movie soundtrack (DTS) to AAC. The average bitrate for that one was 468kbps.